The present invention relates to the field of VoIP telephony and, more particularly, to optimizing the quality of audio within a teleconferencing session via an adaptive codec switching. Contemporary Voice-Over-IP (VoIP) systems typically negotiate a single codec for the entire VoIP session life time of a teleconference. For example, during an initial setup of a VoIP conference call, a VoIP server can select a VoIP audio/video codec which can be utilized based on several criteria such as network conditions, participant quantity, and server settings. This approach provides simple call setup and teardown which can minimize server load. However, as network conditions change, call quality can decrease dramatically. For example, when packet loss between a VoIP client device and the VoIP server increases, the codec chosen by the server can perform poorly resulting in garbled audio which can be incomprehensible to participants using the client device. That is, as different codecs can perform differently well under certain network conditions like delay, jitter and/or packet loss, this can lead to a reduction of quality if network conditions change during the call.
While these codecs can adapt to a limited degree to changing network conditions such as available bandwidth, network delay, and/or packet loss rate change in the meantime, the VoIP clients (e.g., VoIP application) abide with their initial codec choice. Hence, the clients often apply a codec that is not well suited for the present network situation although better codec choices can be available. Additionally, when a teleconference session grows sufficiently large and the number of codecs required for each client device increases, server computation demands can increase drastically. This can cause servers to become unresponsive and utilize lower quality codecs to improve decrease computation, resulting in a poor end user experience of the call.